Use outgoing route/trunk to external IP without registered

HI,
I want to make the call dail 9XXXX then route to outgouing trunk to another IP.

So,in gui ,I create the runk with outgoing setting .

peer detail

host=10.1.1.139
username=
secret=
type=peer
fromdomain=10.1.1.139
allow=alaw&ulaw
qualify=yes
context=outbound-all-routes

IT is no need to register to the external IP 10.1.1.139

in GUI , I add route ,

that mean when I dial 9 , it will use the above trunk …

And in sip.conf

I define the conext of the calling no…

[20001002]
type=peer
host=dynamic
secret=
deny=0.0.0.0/0
permit=10.1.0.0/255.255.0.0
context=outbound-all-routes
insecure=port,invite
fromdomain=10.1.1.139

where 10.1.1.139 is the extenal ip I want to route the call…

But when I make call using 20001002 , the call fail …
in debug I see
<------------->
[Dec 5 20:14:14] VERBOSE[3404] chan_sip.c: — (13 headers 20 lines) —
[Dec 5 20:14:14] VERBOSE[3404] netsock.c: == Using SIP RTP CoS mark 5
[Dec 5 20:14:14] VERBOSE[3404] netsock.c: == Using SIP VRTP CoS mark 6
[Dec 5 20:14:14] VERBOSE[3404] chan_sip.c: Sending to 10.1.1.110 : 5060 (no NAT)
[Dec 5 20:14:14] VERBOSE[3404] chan_sip.c: Using INVITE request as basis request - 11251
[Dec 5 20:14:14] VERBOSE[3404] chan_sip.c: Found peer ‘20001002’ for ‘20001002’ from 10.1.1.110:5060
[Dec 5 20:14:14] VERBOSE[3404] chan_sip.c: Found RTP audio format 111
[Dec 5 20:14:14] VERBOSE[3404] chan_sip.c: Found RTP audio format 110
[Dec 5 20:14:14] VERBOSE[3404] chan_sip.c: Found RTP audio format 3
[Dec 5 20:14:14] VERBOSE[3404] chan_sip.c: Found RTP audio format 0
[Dec 5 20:14:14] VERBOSE[3404] chan_sip.c: Found RTP audio format 8
[Dec 5 20:14:14] VERBOSE[3404] chan_sip.c: Found RTP audio format 112
[Dec 5 20:14:14] VERBOSE[3404] chan_sip.c: Found RTP audio format 101
[Dec 5 20:14:14] VERBOSE[3404] chan_sip.c: Found audio description format speex for ID 111
[Dec 5 20:14:14] VERBOSE[3404] chan_sip.c: Found audio description format speex for ID 110
[Dec 5 20:14:14] VERBOSE[3404] chan_sip.c: Found audio description format speex for ID 112
[Dec 5 20:14:14] VERBOSE[3404] chan_sip.c: Found audio description format telephone-event for ID 101
[Dec 5 20:14:14] VERBOSE[3404] chan_sip.c: Found RTP video format 99
[Dec 5 20:14:14] VERBOSE[3404] chan_sip.c: Found RTP video format 97
[Dec 5 20:14:14] VERBOSE[3404] chan_sip.c: Found RTP video format 98
[Dec 5 20:14:14] VERBOSE[3404] chan_sip.c: Found video description format MP4V-ES for ID 99
[Dec 5 20:14:14] VERBOSE[3404] chan_sip.c: Found video description format H263-1998 for ID 98
[Dec 5 20:14:14] VERBOSE[3404] chan_sip.c: Capabilities: us - 0x1c010e (gsm|ulaw|alaw|g729|h261|h263|h263p), peer - audio=0xa1e (gsm|ulaw|alaw|g726|speex|g726aal2)/video=0x500400 (ilbc|h263p|mpeg4)/text=0x0 (nothing), combined - 0x10000e (gsm|ulaw|alaw|h263p)
[Dec 5 20:14:14] VERBOSE[3404] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Dec 5 20:14:14] VERBOSE[3404] chan_sip.c: Peer audio RTP is at port 10.1.1.110:7078
[Dec 5 20:14:14] VERBOSE[3404] chan_sip.c: Peer video RTP is at port 10.1.1.110:9078
[Dec 5 20:14:14] VERBOSE[3404] chan_sip.c: Looking for 91111111 in outbound-all-routes (domain 10.1.1.89)
[Dec 5 20:14:14] VERBOSE[3404] chan_sip.c:
<— Reliably Transmitting (no NAT) to 10.1.1.110:5060 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.1.1.110:5060;branch=z9hG4bK12184;received=10.1.1.110;rport=5060
From: sip:[email protected];tag=15105
To: sip:[email protected];tag=as1e5a1f1f
Call-ID: 11251
CSeq: 20 INVITE
Server: Asterisk PBX 1.6.2.11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

I can still see the call only route to 10.1.1.89 that is the local ip of the server only …

Dec 5 20:14:14] VERBOSE[3404] chan_sip.c: Looking for 91111111 in outbound-all-routes (domain 10.1.1.89)

I expect it can try to reach 10.1.1.139

So, what do I need to set if I want to make a outgoing trunk and route by dial prefix 9 only and then go to 10.1.1.139 and no need to reigster …

Please advice …

==========
I also find below debug

[Dec 5 23:44:58] VERBOSE[6913] pbx.c: – Executing [[email protected]:19] Dial(“SIP/20001002-00000023”, “SIP/trunk_hmp/992222222,300,”) in new stack
[Dec 5 23:44:58] VERBOSE[6913] netsock.c: == Using SIP RTP CoS mark 5
[Dec 5 23:44:58] VERBOSE[6913] netsock.c: == Using SIP VRTP CoS mark 6
[Dec 5 23:44:58] WARNING[6913] chan_sip.c: No such host: trunk_hmp
[Dec 5 23:44:58] VERBOSE[6913] chan_sip.c: Really destroying SIP dialog ‘[email protected]’ Method: INVITE
[Dec 5 23:44:58] WARNING[6913] app_dial.c: Unable to create channel of type ‘SIP’ (cause 20 - Unknown)
[Dec 5 23:44:58] VERBOSE[6913] app_dial.c: == Everyone is busy/congested at this time (1:0/0/1)
[Dec 5 23:44:58] VERBOSE[6913] pbx.c: – Executing [[email protected]:20] NoOp(“SIP/20001002-00000023”, “Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 20”) in new stack
[Dec 5 23:44:58] VERBOSE[6913] pbx.c: – Executing [[email protected]:21] Goto(“SIP/20001002-00000023”, “s-CHANUNAVAIL,1”) in new stack

The trunk trunk_hmp set to host 10.1.1.139 .The IP is up and can ping to it .
There is nothing send to this IP as this IP open the wireshark …
Why it is still no route to this trunk …

Please advice …

chuikingman - Why are you doing anything in sip.conf? It just gets overwritten by FreePBX.

Also the context you designated is non-existent.

You did not tell us anything about your install, Asterisk or FreePBX version numbers.

Please supply proper information if you want assistance.

Hi,
I only download the astersik now 1.7.1 iso 32 bit file
http://www.asterisk.org/downloads
and install…
everything is default …

from cli
localhostCLI> core show version
Asterisk 1.6.2.11 built by root @ localhost.localdomain on a i686 running Linux on 2010-08-24 20:43:18 UTC
localhost
CLI>

So, how can I make a sip trunk and no need to register to the external IP and route all the outgoing call to this IP ???
Please adivce detailed method ???

Thank

I already told you your contexts are from.

I also asked you what the purpose of editing sip.conf is.

The phrase “Please adivce detailed method” does not make any sense in English but I assume you are asking for step by step instructions. Since we have no idea what device you are trying to talk to that would be difficult to do even if someone wanted to do it.

I strongly suggest you read the documentation on extensions and trunks. Leave your contexts alone.

You should also look at the ‘insecure=port,invite’ directive for your trunk. It disables sip authentication methods.

Hi,
I edit the sip .conf and remove the context field.
[20001002]
type=peer
host=dynamic
secret=
deny=0.0.0.0/0
permit=10.1.0.0/255.255.0.0

it is the calling number .

I want to make the system have below setup.
The callign number 20001002 make call 9XXXXXXX and rech outgoing route and tehn reach the trunk_hmp .
trunk_hmp like below
host=10.1.1.139
type=peer
allow=ulaw,gsm,alaw
outboundproxy=10.1.1.139
trunk=yes
canreinvite=yes
insecure=port,invite

The the call will reach to external IP 10.1.1.139 .The IP do not accept/create register message .It only accept the call go inot this IP …

But in the debug , I see below …
<— SIP read from UDP:10.1.1.110:5060 —>
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.1.1.110:5060;rport;branch=z9hG4bK21454
From: sip:[email protected];tag=19433
To: sip:[email protected]
Call-ID: 17148
CSeq: 20 INVITE
Contact: sip:[email protected]
Content-Type: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Max-Forwards: 70
User-Agent: Linphone/3.4.3 (eXosip2/3.3.0)
Subject: Phone call
Content-Length: 478

v=0
o=20001002 2942 2942 IN IP4 10.1.1.110
s=Talk
c=IN IP4 10.1.1.110
t=0 0
m=audio 7078 RTP/AVP 111 110 3 0 8 112 101
a=rtpmap:111 speex/16000
a=fmtp:111 vbr=on
a=rtpmap:110 speex/8000
a=fmtp:110 vbr=on
a=rtpmap:112 speex/32000
a=fmtp:112 vbr=on
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
m=video 9078 RTP/AVP 99 97 98
a=rtpmap:99 MP4V-ES/90000
a=fmtp:99 profile-level-id=3
a=rtpmap:97 theora/90000
a=rtpmap:98 H263-1998/90000
a=fmtp:98 CIF=1;QCIF=1

<------------->
[Dec 6 17:00:24] VERBOSE[3488] chan_sip.c: — (13 headers 20 lines) —
[Dec 6 17:00:24] VERBOSE[3488] netsock.c: == Using SIP RTP CoS mark 5
[Dec 6 17:00:24] VERBOSE[3488] netsock.c: == Using SIP VRTP CoS mark 6
[Dec 6 17:00:24] VERBOSE[3488] chan_sip.c: Sending to 10.1.1.110 : 5060 (no NAT)
[Dec 6 17:00:24] VERBOSE[3488] chan_sip.c: Using INVITE request as basis request - 17148
[Dec 6 17:00:24] VERBOSE[3488] chan_sip.c: Found peer ‘20001002’ for ‘20001002’ from 10.1.1.110:5060
[Dec 6 17:00:24] VERBOSE[3488] chan_sip.c: Found RTP audio format 111
[Dec 6 17:00:24] VERBOSE[3488] chan_sip.c: Found RTP audio format 110
[Dec 6 17:00:24] VERBOSE[3488] chan_sip.c: Found RTP audio format 3
[Dec 6 17:00:24] VERBOSE[3488] chan_sip.c: Found RTP audio format 0
[Dec 6 17:00:24] VERBOSE[3488] chan_sip.c: Found RTP audio format 8
[Dec 6 17:00:24] VERBOSE[3488] chan_sip.c: Found RTP audio format 112
[Dec 6 17:00:24] VERBOSE[3488] chan_sip.c: Found RTP audio format 101
[Dec 6 17:00:24] VERBOSE[3488] chan_sip.c: Found audio description format speex for ID 111
[Dec 6 17:00:24] VERBOSE[3488] chan_sip.c: Found audio description format speex for ID 110
[Dec 6 17:00:24] VERBOSE[3488] chan_sip.c: Found audio description format speex for ID 112
[Dec 6 17:00:24] VERBOSE[3488] chan_sip.c: Found audio description format telephone-event for ID 101
[Dec 6 17:00:24] VERBOSE[3488] chan_sip.c: Found RTP video format 99
[Dec 6 17:00:24] VERBOSE[3488] chan_sip.c: Found RTP video format 97
[Dec 6 17:00:24] VERBOSE[3488] chan_sip.c: Found RTP video format 98
[Dec 6 17:00:24] VERBOSE[3488] chan_sip.c: Found video description format MP4V-ES for ID 99
[Dec 6 17:00:24] VERBOSE[3488] chan_sip.c: Found video description format H263-1998 for ID 98
[Dec 6 17:00:24] VERBOSE[3488] chan_sip.c: Capabilities: us - 0x1c010e (gsm|ulaw|alaw|g729|h261|h263|h263p), peer - audi o=0xa1e (gsm|ulaw|alaw|g726|speex|g726aal2)/video=0x500400 (ilbc|h263p|mpeg4)/text=0x0 (nothing), combined - 0x10000e (gs m|ulaw|alaw|h263p)
[Dec 6 17:00:24] VERBOSE[3488] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (teleph one-event), combined - 0x1 (telephone-event)
[Dec 6 17:00:24] VERBOSE[3488] chan_sip.c: Peer audio RTP is at port 10.1.1.110:7078
[Dec 6 17:00:24] VERBOSE[3488] chan_sip.c: Peer video RTP is at port 10.1.1.110:9078
[Dec 6 17:00:24] VERBOSE[3488] chan_sip.c: Looking for 92222222 in default (domain 10.1.1.89)
[Dec 6 17:00:24] VERBOSE[3488] chan_sip.c:
<— Reliably Transmitting (no NAT) to 10.1.1.110:5060 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.1.1.110:5060;branch=z9hG4bK21454;received=10.1.1.110;rport=5060
From: sip:[email protected];tag=19433
To: sip:[email protected];tag=as06a7770f
Call-ID: 17148
CSeq: 20 INVITE
Server: Asterisk PBX 1.6.2.11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

<------------>
[Dec 6 17:00:24] NOTICE[3488] chan_sip.c: Call from ‘20001002’ to extension ‘92222222’ rejected because extension not fo und in context ‘default’.
[Dec 6 17:00:24] VERBOSE[3488] chan_sip.c: Scheduling destruction of SIP dialog ‘17148’ in 32000 ms (Method: INVITE)
[Dec 6 17:00:24] VERBOSE[3488] chan_sip.c:
<— SIP read from UDP:10.1.1.110:5060 —>
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.1.1.110:5060;rport;branch=z9hG4bK21454
Route: sip:10.1.1.89;lr
From: sip:[email protected];tag=19433
To: sip:[email protected];tag=as06a7770f
Call-ID: 17148
CSeq: 20 ACK
Content-Length: 0

<------------->
[Dec 6 17:00:24] VERBOSE[3488] chan_sip.c: — (8 headers 0 lines) —
[Dec 6 17:00:24] VERBOSE[3488] chan_sip.c: Really destroying SIP dialog ‘17148’ Method: ACK
[Dec 6 17:00:34] VERBOSE[3488] chan_sip.c:
<— SIP read from UDP:10.1.1.110:5060 —>
jaK
<------------->

It seem the call invite 922222222 in domain 10.1.1.89 only , do not route the call to reach ip 10.1.1.139 …

Please advice how I can setup to route the call to reach IP 10.1.1.139 detail…

For the third time I will tell you not to edit the sip.conf and setup a freepbx extension.

Second, the answer to the question is right in your log:

Change the context of your trunk to from-trunk and make sure you have an inbound route setup or from-internal and have a matching extension.

I only need to route to external IP 10.1.1.139.
Not withing the domain 10.1.1.89 …

The 9XXXXXX is in external IP 10.1.1.139 …
it is outbound call…

Please advice again…

Hi,
I have below sip.conf
[20001002]
type=peer
host=dynamic
secret=
deny=0.0.0.0/0
permit=10.2.0.0/255.255.0.0
context=from-internal

I have below trunk peer detail
host=10.2.1.51
type=peer
allow=ulaw,gsm,alaw
outboundproxy=10.2.1.51
trunk=yes
canreinvite=yes
insecure=port,invite
context=from-trunk

I make call from 20001002 to 92222221 .
I expect the call will route to outgoing route and go to trunk “trunk_hmp” and that trunk to go to ip 10.2.1.51 .That is a application to receive the call.
The application at ip 10.2.1.51 is no need to register in the FPBX and it do not accept registration message .

I paste the debug and please advice what is wrong …

[[email protected] ~]# asterisk -rvv
Asterisk 1.6.2.11, Copyright © 1999 - 2010 Digium, Inc. and others.
Created by Mark Spencer [email protected]
Asterisk comes with ABSOLUTELY NO WARRANTY; type ‘core show warranty’ for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type ‘core show license’ for details.

== Parsing ‘/etc/asterisk/asterisk.conf’: == Found
Connected to Asterisk 1.6.2.11 currently running on localhost (pid = 3445)
Verbosity is at least 3

<— SIP read from UDP:10.2.1.61:5060 —>
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.2.1.61:5060;rport;branch=z9hG4bK8205
From: sip:[email protected];tag=14371
To: sip:[email protected]
Call-ID: 16637
CSeq: 20 INVITE
Contact: sip:[email protected]
Content-Type: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Max-Forwards: 70
User-Agent: Linphone/3.4.3 (eXosip2/3.3.0)
Subject: Phone call
Content-Length: 541

v=0
o=20001002 1446 1446 IN IP4 10.2.1.61
s=Talk
c=IN IP4 10.2.1.61
t=0 0
m=audio 7078 RTP/AVP 112 111 110 3 0 8 101
a=rtpmap:112 speex/32000
a=fmtp:112 vbr=on
a=rtpmap:111 speex/16000
a=fmtp:111 vbr=on
a=rtpmap:110 speex/8000
a=fmtp:110 vbr=on
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
m=video 9078 RTP/AVP 102 99 97 98
a=rtpmap:102 H264/90000
a=fmtp:102 profile-level-id=428014
a=rtpmap:99 MP4V-ES/90000
a=fmtp:99 profile-level-id=3
a=rtpmap:97 theora/90000
a=rtpmap:98 H263-1998/90000
a=fmtp:98 CIF=1;QCIF=1

<------------->
— (13 headers 22 lines) —
== Using SIP RTP CoS mark 5
== Using SIP VRTP CoS mark 6
Sending to 10.2.1.61 : 5060 (no NAT)
Using INVITE request as basis request - 16637
Found peer ‘20001002’ for ‘20001002’ from 10.2.1.61:5060
Found RTP audio format 112
Found RTP audio format 111
Found RTP audio format 110
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format speex for ID 112
Found audio description format speex for ID 111
Found audio description format speex for ID 110
Found audio description format telephone-event for ID 101
Found RTP video format 102
Found RTP video format 99
Found RTP video format 97
Found RTP video format 98
Found video description format H264 for ID 102
Found video description format MP4V-ES for ID 99
Found video description format H263-1998 for ID 98
Capabilities: us - 0x1c010e (gsm|ulaw|alaw|g729|h261|h263|h263p), peer - audio=0xa1e (gsm|ulaw|alaw|g726|speex|g726aal2)/video=0x700400 (ilbc|h263p|h264|mpeg4)/text=0x0 (nothing), combined - 0x10000e (gsm|ulaw|alaw|h263p)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.2.1.61:7078
Peer video RTP is at port 10.2.1.61:9078
Looking for 922222221 in from-internal (domain 10.2.1.102)
list_route: hop: sip:[email protected]

<— Transmitting (no NAT) to 10.2.1.61:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.2.1.61:5060;branch=z9hG4bK8205;received=10.2.1.61;rport=5060
From: sip:[email protected];tag=14371
To: sip:[email protected]
Call-ID: 16637
CSeq: 20 INVITE
Server: Asterisk PBX 1.6.2.11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: sip:[email protected]
Content-Length: 0

<------------>

<— SIP read from UDP:10.2.1.61:5060 —>
jaK
<------------->
– Executing [[email protected]:1] Macro(“SIP/20001002-00000019”, “user-callerid,SKIPTTL,”) in new stack
– Executing [[email protected]:1] Set(“SIP/20001002-00000019”, “AMPUSER=20001002”) in new stack
– Executing [[email protected]:2] GotoIf(“SIP/20001002-00000019”, “0?report”) in new stack
– Executing [[email protected]:3] ExecIf(“SIP/20001002-00000019”, “1?Set(REALCALLERIDNUM=20001002)”) in new stack
– Executing [[email protected]:4] Set(“SIP/20001002-00000019”, “AMPUSER=20001002”) in new stack
– Executing [[email protected]:5] Set(“SIP/20001002-00000019”, “AMPUSERCIDNAME=20001002”) in new stack
– Executing [[email protected]:6] GotoIf(“SIP/20001002-00000019”, “0?report”) in new stack
– Executing [[email protected]:7] Set(“SIP/20001002-00000019”, “AMPUSERCID=20001002”) in new stack
– Executing [[email protected]:8] Set(“SIP/20001002-00000019”, “CALLERID(all)=“20001002” <20001002>”) in new stack
– Executing [[email protected]:9] GotoIf(“SIP/20001002-00000019”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,18)
– Executing [[email protected]:18] NoOp(“SIP/20001002-00000019”, “Using CallerID “20001002” <20001002>”) in new stack
– Executing [[email protected]:2] NoCDR(“SIP/20001002-00000019”, “”) in new stack
– Executing [[email protected]:3] Wait(“SIP/20001002-00000019”, “1”) in new stack
– Executing [[email protected]:4] Playback(“SIP/20001002-00000019”, “silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer”) in new stack
– <SIP/20001002-00000019> Playing ‘silence/1.gsm’ (language ‘en’)
– <SIP/20001002-00000019> Playing ‘cannot-complete-as-dialed.gsm’ (language ‘en’)
– <SIP/20001002-00000019> Playing ‘check-number-dial-again.gsm’ (language ‘en’)
– Executing [[email protected]:5] Wait(“SIP/20001002-00000019”, “1”) in new stack
– Executing [[email protected]:6] Congestion(“SIP/20001002-00000019”, “20”) in new stack

<— Reliably Transmitting (no NAT) to 10.2.1.61:5060 —>
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 10.2.1.61:5060;branch=z9hG4bK8205;received=10.2.1.61;rport=5060
From: sip:[email protected];tag=14371
To: sip:[email protected];tag=as704ed5fa
Call-ID: 16637
CSeq: 20 INVITE
Server: Asterisk PBX 1.6.2.11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

<------------>
== Spawn extension (from-internal, 922222221, 6) exited non-zero on ‘SIP/20001002-00000019’
– Executing [[email protected]:1] Macro(“SIP/20001002-00000019”, “hangupcall”) in new stack
– Executing [[email protected]:1] GotoIf(“SIP/20001002-00000019”, “1?skiprg”) in new stack
– Goto (macro-hangupcall,s,4)
– Executing [[email protected]:4] GotoIf(“SIP/20001002-00000019”, “1?skipblkvm”) in new stack
– Goto (macro-hangupcall,s,7)
– Executing [[email protected]:7] GotoIf(“SIP/20001002-00000019”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,9)
– Executing [[email protected]:9] Hangup(“SIP/20001002-00000019”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on ‘SIP/20001002-00000019’ in macro ‘hangupcall’
== Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/20001002-00000019’

<— SIP read from UDP:10.2.1.61:5060 —>
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.2.1.61:5060;rport;branch=z9hG4bK8205
From: sip:[email protected];tag=14371
To: sip:[email protected];tag=as704ed5fa
Call-ID: 16637
CSeq: 20 ACK
Content-Length: 0

<------------->
— (7 headers 0 lines) —
Really destroying SIP dialog ‘16637’ Method: ACK

<— SIP read from UDP:10.2.1.61:5060 —>
jaK
<------------->
localhost*CLI> quit
Executing last minute cleanups
[[email protected] ~]#

Your question makes no sense.